The Future of Open Source VoIP is Asterisk 1.6

Maintainer of Asterisk speaks to the faithful and tells them the future is very bright.

TORONTO -- Asterisk hasn't been around all that long, but it has become the standard for open source VoIP PBX .

In a keynote session at the Asterisk and Open Telephony Conference here, Kevin Fleming, senior software engineer at Digium and co-maintainer of Asterisk, told an overflow capacity crowd of Asterisk lovers what the future holds for their beloved project.

With the 1.4 release now under his belt, as well as a new Asterisk appliance effort called AsteriskNOW, Fleming is now concentrating his efforts on the 1.6 release, which could be out by the end of this year.

As part of the development process, Asterisk developers have overhauled how they deal with security issues. Fleming boasted that security issues are handled by a core team that address issues 24 to 48 hours from the initial report.

There is also now a formal Asterisk Project Security Advisory reporting system that will provide full disclosure into reported issues and what Asterisk is doing to fix issues.

Fleming admitted that Asterisk hasn't done a great job of making users aware of security advisories, but the new effort is an attempt to do better.

"One of the benefits of open source is that there are more people using our code and a lot more people looking at our code and telling us when things are wrong," Fleming said.

"No one gets to look at the code for Nortel's systems and there are probably just as many vulnerabilities there but you only hear about it when it gets exploited."

In addition to the improved security vulnerabilities, Fleming said that the Asterisk community is working on an effort called Asterisk Network Edge, which will enable an Asterisk deployment to better integrate with security devices such as firewalls.

In terms of actual new features that will land in the upcoming 1.6 Asterisk, Fleming highlighted additional Call Bridging functionality as a key deliverable.

The goal for 1.6 is to provide flexible multi-channel bridging such that channels that can be added and dropped so calls can instantly be converted from two party to multi-party. That functionality will enable the use of advance features, such as in call announcements and call center monitoring, by a third party.

Improved call handling capacity and performance is also set for the 1.6 release. Fleming explained that the idea is to reduce the thread/memory footprint for handling large numbers of media streams. Initial tests already show a 100 to 200 percent improvement in the number of streams that a server can handle.

Support for asynchronous events will also appear in 1.6, enabling modules in Asterisk to communicate with each other across a cluster. Fleming told the audience that, with current Asterisk versions, voice mail needs to be on the same server as your call is on in order for call notification to actually work.

Better reporting through a new call event logging capability in Asterisk 1.6 will allow complete tracking of events that take place during a call. The goal, according to Fleming, is to provide more detail than traditional CDR (Call Detail Recording) features offer and to allow for more granular tracking and auditing.

Looking beyond Asterisk 1.6, Fleming noted that there are a few high-priority items on the list to tackle, including clustering and failover support.

Even with all the features that Fleming has on his roadmap for Asterisk, he noted that he can't be sure at this point what will end up in future versions.

"We can't predict that," Fleming said. "It will be whatever people tell us they want and contribute."

Fleming said the current demands placed on Asterisk by users are a far cry from where Asterisk began from, and noted that it's difficult for him to talk about where Asterisk may be going in the future since the very first release of Asterisk that was intended to be used by anyone was only 2.5 years ago.

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